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dad91a69bf
Today, behaviour is decided based on if transport sequence number v2 is in the SDP answer. But it might be better to decide based on received packets since it is valid to negotiate both extensions. Another bonus With this solution is that Call does not need to know about receive header exensions. This is an alternative to https://webrtc-review.googlesource.com/c/src/+/291337 Bug: webrtc:7135 Change-Id: Ib75474127d6e2e2029557b8bb2528eaac66979f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291525 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39226}
128 lines
5.1 KiB
C++
128 lines
5.1 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_
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#define MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_
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#include <deque>
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#include <functional>
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#include <memory>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/field_trials_view.h"
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#include "api/rtp_headers.h"
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#include "api/transport/network_control.h"
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#include "api/units/data_size.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "modules/remote_bitrate_estimator/packet_arrival_map.h"
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#include "modules/rtp_rtcp/source/rtcp_packet.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "rtc_base/numerics/sequence_number_unwrapper.h"
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#include "rtc_base/synchronization/mutex.h"
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namespace webrtc {
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// Class used when send-side BWE is enabled: This proxy is instantiated on the
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// receive side. It buffers a number of receive timestamps and then sends
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// transport feedback messages back too the send side.
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class RemoteEstimatorProxy {
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public:
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// Used for sending transport feedback messages when send side
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// BWE is used.
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using TransportFeedbackSender = std::function<void(
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std::vector<std::unique_ptr<rtcp::RtcpPacket>> packets)>;
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RemoteEstimatorProxy(TransportFeedbackSender feedback_sender,
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NetworkStateEstimator* network_state_estimator);
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~RemoteEstimatorProxy();
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void IncomingPacket(const RtpPacketReceived& packet);
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// TODO(perkj, bugs.webrtc.org/14859): Remove all usage. This method is
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// currently not used by PeerConnections.
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void IncomingPacket(int64_t arrival_time_ms,
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size_t payload_size,
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const RTPHeader& header);
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// Sends periodic feedback if it is time to send it.
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// Returns time until next call to Process should be made.
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TimeDelta Process(Timestamp now);
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void OnBitrateChanged(int bitrate);
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void SetTransportOverhead(DataSize overhead_per_packet);
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private:
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struct Packet {
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Timestamp arrival_time;
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DataSize size;
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uint32_t ssrc;
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absl::optional<uint32_t> absolute_send_time_24bits;
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absl::optional<uint16_t> transport_sequence_number;
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absl::optional<FeedbackRequest> feedback_request;
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};
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void IncomingPacket(Packet packet) RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
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void MaybeCullOldPackets(int64_t sequence_number, Timestamp arrival_time)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
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void SendPeriodicFeedbacks() RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
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void SendFeedbackOnRequest(int64_t sequence_number,
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const FeedbackRequest& feedback_request)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
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// Returns a Transport Feedback packet with information about as many packets
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// that has been received between [`begin_sequence_number_incl`,
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// `end_sequence_number_excl`) that can fit in it. If `is_periodic_update`,
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// this represents sending a periodic feedback message, which will make it
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// update the `periodic_window_start_seq_` variable with the first packet that
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// was not included in the feedback packet, so that the next update can
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// continue from that sequence number.
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//
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// If no incoming packets were added, nullptr is returned.
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//
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// `include_timestamps` decide if the returned TransportFeedback should
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// include timestamps.
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std::unique_ptr<rtcp::TransportFeedback> MaybeBuildFeedbackPacket(
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bool include_timestamps,
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int64_t begin_sequence_number_inclusive,
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int64_t end_sequence_number_exclusive,
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bool is_periodic_update) RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
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const TransportFeedbackSender feedback_sender_;
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Timestamp last_process_time_;
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Mutex lock_;
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// `network_state_estimator_` may be null.
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NetworkStateEstimator* const network_state_estimator_
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RTC_PT_GUARDED_BY(&lock_);
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uint32_t media_ssrc_ RTC_GUARDED_BY(&lock_);
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uint8_t feedback_packet_count_ RTC_GUARDED_BY(&lock_);
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SeqNumUnwrapper<uint16_t> unwrapper_ RTC_GUARDED_BY(&lock_);
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DataSize packet_overhead_ RTC_GUARDED_BY(&lock_);
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// The next sequence number that should be the start sequence number during
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// periodic reporting. Will be absl::nullopt before the first seen packet.
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absl::optional<int64_t> periodic_window_start_seq_ RTC_GUARDED_BY(&lock_);
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// Packet arrival times, by sequence number.
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PacketArrivalTimeMap packet_arrival_times_ RTC_GUARDED_BY(&lock_);
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TimeDelta send_interval_ RTC_GUARDED_BY(&lock_);
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bool send_periodic_feedback_ RTC_GUARDED_BY(&lock_);
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// Unwraps absolute send times.
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uint32_t previous_abs_send_time_ RTC_GUARDED_BY(&lock_);
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Timestamp abs_send_timestamp_ RTC_GUARDED_BY(&lock_);
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};
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} // namespace webrtc
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#endif // MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_
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